This invention relates in general to data transmission networks and in particular to a method for controlling the loading of internet protocol transmission networks.
Telephone systems have become increasingly sophisticated and are designed to provide many services to subscribers. A typical conventional telephone network 10, also referred to as a switched circuit network, is schematically illustrated in FIG. 1 where the arrows indicate the flow of information within the network. The network 10 has been simplified to illustrate its operation. An individual subscriber telephone 12 is connected to a Local Telephone Company, or Local Exchange Carrier (LEC) 13 by a Service Switching Point (SSP) 14. While one telephone 12 is shown in FIG. 1, it will be appreciated that a plurality of telephones and/or other devices, such as personal computers also may be connected from the subscriber to the SSP 14.
The SSP 14 provides a gateway for connection to long distance carriers, wireless networks and other local telephone companies which are collectively shown in FIG. 1 as a Public Switched Telephone Network (PSTN) 16. The SSP 14 is also connected through a Signaling Transfer Point (STP) 17 to a Service Control Point (SCP) 18. The STP 17 functions as a signal router while the SCP 18 includes a data base and operating instructions for the SSP 14. As shown by the double headed arrows in FIG. 1, information flows in both directions between the individual components. The connections between the SSP 14, the STP 17 and the SCP 18 are indicated by dashed lines. The dashed lines represent signaling paths between the SSP 14, the STP 17 and the SCP 18 for digital control signals. These signaling paths are not voice bearing paths, but are reserved for the control signals. Logic contained in the SCP 18 responds to digital request signals sent through the STP 17 with instructions concerning how the LEC switch 14 should respond to both incoming and outgoing calls.
The digital control signals are defined by the Common Signaling System No. 7, or SS7, which is a global standard for telecommunications defined by the International Telephone Union (ITU). The SS7 standard defines the procedures and protocol by which network elements in the PSTN exchange information over a digital signaling network to effect wireless, or cellular, and wireline call setup, routing, control and teardown. The SS7 messages are exchanged between network elements over bi-directional channels called signaling links which are also shown as dashed lines and labeled as such in FIG. 1. Thus, the SS7 signaling and protocol occurs out-of-band on dedicated channels instead of in-band on voice channels, such as the solid line labeled voice trunk in FIG. 1. The SS7 protocol provides both faster call set up times and more efficient use of the available voice circuits. Once a call has been arranged with the SS7 protocol, actual voice communication is established over voice “bearer” lines, contained in groups called voice trunks, one of which is shown in FIG. 1 with a solid line.
A second set of components are shown on the right side of FIG. 1 and represent a second SSP 20 that is contained in a second Local Exchange Carrier, or LEC, 21. The second SSP 20 is connected to another subscriber telephone 22. Also included in the right portion of FIG. 2 are a second STP 24 and a second SCP 26. The various SS7 links between the two STP's 17 and 24 and the two SCP's 18 and 26 are shown in FIG. 1 passing through the PSTN 16, as does the voice trunk line. It will be appreciated that both the SS7 control signals and the voice signal may pass through a number of conventional components within the PSTN 16 that have been omitted from FIG. 1 for simplicity. However, the SS7 components of the first local telephone company communicate with corresponding units in another local telephone company to establish a communication link between the two telephones 12 and 22 via an available voice link. While communication has been illustrated between two local telephone companies in FIG. 1, it will be appreciated that the second STP and SCP also can be included in the same local company (not shown) with communication being established without routing through the PSTN 16.
To illustrate the operation of the network 10, assume that the subscriber with the first telephone 12 desires to call the subscriber 22 with the second telephone having an out-of-switch number, that is, the second telephone 22 is not connected to the same SSP 14 as the first telephone 12. The originating SSP 14 transmits an Integrated Services digital network User Part (ISUP) Initial Address Message (IAM) to the first STP 17 to reserve an idle voice circuit from the originating SSP 14 to the destination SSP 20. The IAM includes the originating point code, the destination point code, the voice trunk identification code, dialed digits and, optionally, the calling party number. The IAM is routed over a signaling link from the first STP 17 to the destination SSP 20. The destination SSP 20 determines that it serves the called telephone 22 and that the line is available for ringing. The destination SSP 20 rings the called party line and transmits an ISUP Address complete Message (ACM) via the second STP 24 to the originating SSP 14 to indicate that the remote end of the voice trunk has been reserved. When the second subscriber picks up his telephone 22, the destination SSP 20 terminates the ringing tone and transmits an ISUP Answer Message (ANM) to the originating SSP 14 via its home STP 24. The originating SSP 14 verifies that the calling party's line is connected to the voice trunk and, if connected, initiates billing. If the calling party hangs up first, the originating SSP 14 sends an ISUP Release Message (REL) to the destination SSP 20 to release the voice trunk via associated STPs 17 and 24, respectively. If the called party hangs up first, or if the line is busy, the second SSP 20 sends a REL to the originating SSP 14, again via associated STPs 24 and 17, respectively. Upon receiving the REL from the originating SSP 14, the destination SSP 20 disconnects the voice trunk from the called party's line and sets the voice trunk to idle. The destination SSP 20 then transmits an ISUP Release Complete Message (RLC) to the originating SSP 14. When the originating SSP 14 receives the RLC, it terminates the billing cycle and sets the voice trunk to idle in preparation for the next call.
The development of the internet has further enhanced the telephonic communications with the concurrent development of a Voice-over-Internet Protocol (VoIP) Telephone companies have found that it is sometimes cheaper to carry voice traffic over Internet Protocol (IP) networks than over traditional switched circuit networks because an IP telephony network can make better use of available bandwidth. In a VoIP network, digitized voice data is highly compressed and carried in packets over IP networks, which are commonly referred to as “backbone networks”. Using the same bandwidth, a VoIP network can carry many times the number of voice calls as a switched circuit network. The use of VoIP networks has been so successful that most telecommunications companies have established dedicated backbone networks to provide VoIP service to their customers.
A typical VoIP network 30 is illustrated in FIG. 2 by a simplified schematic diagram. Components shown in FIG. 2 that are similar to components shown in FIG. 1 have the same numerical identifiers. To the left of the diagram is the Local Exchange Company (LEC) 13 shown in the left portion of FIG. 1; however, in FIG. 2, the LEC is connected to an IP backbone network 32. It will be appreciated that the LEC 13 would also be connected to the Public Switched Telephone Network 16, as shown in FIG. 1; however, for simplicity, the connection to the Public Switched Telephone Network 16 is not shown in FIG. 2. As shown in FIG. 2, the subscriber has a telephone 12 connected to the SSP 14. The SSP 14 is connected through a media gateway 34 to the backbone 32. The media gateway 34 receives voice calls and compresses and packetizes the voice data. The packetized voice data is then delivered to the backbone network 32 for transmission to a destination media gateway (not shown). The destination media gateway converts the packetized voice data back to a voice format for transmission to a called party through a destination SSP (not shown). It is also possible to connect phones, such as device 12 shown in FIG. 2, directly to “soft switches” that are composed of elements similar to 34, 36 and 38 and a call control component (not shown) in lieu of connecting to a traditional SSP, as shown in FIG. 2.
As also shown in FIG. 2, the SSP 14 is connected through a corresponding STP 17 to a SCP 18. The SSP 14 is operative to generate SS7 ISUP control messages to set-up and tear-down calls, as described above, while the SCP 18 replies to Transaction Capabilities Application Part (TCAP) query messages to support the exchange of call logic related data across the SS7 portion of the network. However, the STP 17 and SCP 18 are connected to a signaling gateway 36. The signaling gateway 36 converts the SS7 control messages into SS7 IP packets that are relayed through the backbone network 32 to a signaling gateway (not shown) at the destination that converts SS7 IP packets back to SS7 signals. The destination gateway is connected to a destination STP and a destination SCP (not shown) that are operative, upon receiving the SS7 control messages to complete the call, also as described above. Additionally, a media gateway controller 38 is shown in FIG. 2 that is connected to the media gateway 34 and the signaling gateway 36. The SSP 14 exchanges ISUP messages with SSP's via the signaling gateway 36 to handle the registration and management of resources, such as data paths, at the media gateway 34. The media gateway 34, signaling gateway 36 and media gateway controller 38 may be provided by the LEC 13, or the IP backbone network 32, and provide an interface between the conventional SSP 14, STP 17 and SCP 18 and the IP backbone network 32.
While the use of IP backbone networks have enabled telephone companies to increase capacity, the continuing proliferation of service requirements has steadily increased the need for more capacity. Typical native Internet Protocol (IP) service proves a “best effort” service environment since IP, by itself, does not limit requests or manage data packet flow. Thus, the resulting provided services are equally good or poor, depending upon the infrastructure investment made by the provider and the existing network loading conditions. Native IP without traffic engineering is analogous to a busy expressway without provision of traffic stop lights on expressway on-ramps to control vehicle entry onto the expressway. By controlling the rate of vehicles entering the expressway, overloading is avoided, allowing the traffic on the expressway to flow smoothly and rapidly. Without the entrance control signals, the expressway experiences best effort flow and can become overloaded with resulting traffic jams and slow downs. Such best effort flow is not desirable for high priority traffic, such as ambulances, tow trucks and police vehicles, just as best efforts are not desirable for high priority communications.
Typically, telephony service providers have elected to provide equally good service to all consumers regardless of the value they derive and the price they are willing to pay. The net result is very heavy investment by the service provider in infrastructure with all consumers benefiting equally from a high quality of service while all consumers pay in accordance with the lowest consumer value derived. However, some customers desire and/or require a higher service quality, or a lower cost. For example, medical services may desire greater availability of communication lines and/or increased bandwidth to allow consultations between remote locations, such as hospitals in different parts of the country. Providing such enhanced services in a best effort service environment would require further investment in the infrastructure to meet the demand by raising the current service level for all customers. Alternately, separate infrastructure could be added and dedicated to provide the enhanced services. Either of these approaches would be very expensive. Additionally, the first approach would not be required for all of the customers. Accordingly, it would be desirable to provide an alternate approach that would provide a higher level of service to selected customers when desired without having to increase infrastructure investment for all of the involved parties.